Asterisk 11 And Old Thomson 2030S Hardphone => SIP Register/Auth Problem Against V11

Home » Asterisk Users » Asterisk 11 And Old Thomson 2030S Hardphone => SIP Register/Auth Problem Against V11
Asterisk Users 5 Comments

Hi!
I wish you all e Happy New Year first!

Allthough, I’m relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. 🙂
Platform is Debian 8/Asterisk Packages (11) from Debian Repo.

But I am running into problems setting up 2 older Hardphones, Thomson
2030S. 🙁

with in my sip.conf, I have got for this hardphone:
[…]
[hard1]
username=hard1
secret=correct-and-three-times-checked-4-digit-pin
allow=ulaw
allow=alaw
context=internal
type=friend
nat=force_rport,comedia
qualify=yes
host=dynamic
[…]

The SIP debug Output from Asteris CLI :
gw*CLI> sip reload Reloading SIP
== Parsing ‘/etc/asterisk/sip.conf’: Found
== Using SIP CoS mark 4

Here Rebooting/Power up Thomson 2030S on 192.168.11.72

<--- SIP read from UDP:192.168.11.72:5060 --->
REGISTER sip:192.168.11.251;user=phone SIP/2.0
v: SIP/2.0/UDP 192.168.11.72:5060;branch=z9hG4bK8030819752698542643-155044
f: ;tag=c0a80101-25da4
t:
i: 5789b-c0a80101-5-4@192.168.11.72
CSeq: 1 REGISTER
Route:
Max-Forwards: 70
Expires: 60
m:
User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-5A-AA
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold l: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.11.72:5060 (no NAT)
Sending to 192.168.11.72:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.72:5060 --->
SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP
192.168.11.72:5060;branch=z9hG4bK8030819752698542643-155044;received=192.168.11.72
From: ;tag=c0a80101-25da4
To: ;tag=as2220827f Call-ID: 5789b-c0a80101-5-4@192.168.11.72
CSeq: 1 REGISTER
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=”gw”, nonce=”47287142″
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5789b-c0a80101-5-4@192.168.11.72’
in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog ‘5789b-c0a80101-5-4@192.168.11.72’
in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.11.72:5060 --->
REGISTER sip:192.168.11.251;user=phone SIP/2.0
v: SIP/2.0/UDP 192.168.11.72:5060;branch=z9hG4bK1414692036426970487-155047
f: ;tag=c0a80101-25da4
t:
i: 5789b-c0a80101-5-4@192.168.11.72
CSeq: 2 REGISTER
Route:
Max-Forwards: 70
Expires: 60
m:
Authorization: Digest username=”hard1″, realm=”gw”, nonce=”47287142″, uri=”sip:192.168.11.251″, response=”f57f93a5a59e8f72aad5a5a43d19d3bf”, algorithm=MD5
User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-5A-AA
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold l: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.11.72:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.72:5060 --->
SIP/2.0 403 Forbidden Via: SIP/2.0/UDP
192.168.11.72:5060;branch=z9hG4bK1414692036426970487-155047;received=192.168.11.72
From: ;tag=c0a80101-25da4
To: ;tag=as2220827f Call-ID: 5789b-c0a80101-5-4@192.168.11.72
CSeq: 2 REGISTER
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5789b-c0a80101-5-4@192.168.11.72’
in 32000 ms (Method: REGISTER)

Any Ideas to get those Phone working?

TIA / with kind Regards from an Sowy Norther Germany

mit freundlichen Grüßen Jürgen Sauer
—
Jürgen Sauer – automatiX GmbH,
+49-4209-4699, juergen.sauer@automatix.de Geschäftsführer: Jürgen Sauer, Gerichtstand: Amtsgericht Walsrode • HRB 120986
Ust-Id: DE191468481 • St.Nr.: 36/211/08000
GPG Public Key zur Signaturprüfung:
http://www.automatix.de/juergen_sauer_publickey.gpg

—

5 thoughts on - Asterisk 11 And Old Thomson 2030S Hardphone => SIP Register/Auth Problem Against V11

  • In most cases, there is no need to set the “username=” option. The name of the device is the name within the square brackets above the configuration section. Delete the “username=hard1” and reload sip.conf.

  • Am 07.01.2016 um 10:55 schrieb Frank:
    Thx, 4answer. 🙂

    Should be so, agreed. But it worked quite a long time not this way.
    🙁

    Got now up. Why? I do not know. This Hard phone is really needing an full expert”.

    Now this piece of antique hardware it does recognize calls, which asterisk sends. Calling out, works, asterisk sees the device as “hard1”. Calling “hard1”
    shows up, “not avaible”… Same Setup on Snom 821 works perfectly.

    mit freundlichen Grüßen Jürgen Sauer
    —
    Jürgen Sauer – automatiX GmbH,
    +49-4209-4699, juergen.sauer@automatix.de Geschäftsführer: Jürgen Sauer, Gerichtstand: Amtsgericht Walsrode • HRB 120986
    Ust-Id: DE191468481 • St.Nr.: 36/211/08000
    GPG Public Key zur Signaturprüfung:
    http://www.automatix.de/juergen_sauer_publickey.gpg

    —

  • Am 11.01.2016 um 14:21 schrieb Scott Griepentrog:

    Hardware Information HW version V3

    Software Information Boot Code version V1.01
    DSP version V1.00 4 way. App version V1.66

    It seems to be, that this fw can not deal with not-numeric-sip accounts.

    I entered the extension number as name, account and it works. I presume, the sip implemetation is quite old and won’t be upgradeable.

    So, it seem to wor correct.

    BTW: Non Numeric made calls are not accepted by the phone, but callin as pure number ist works.

    Took quite a while to figuere it out 🙂

    Solved by my self, using Try-and-error Metodic. 🙂

    mit freundlichen Grüßen Jürgen Sauer
    —
    Jürgen Sauer – automatiX GmbH,
    +49-4209-4699, juergen.sauer@automatix.de Geschäftsführer: Jürgen Sauer, Gerichtstand: Amtsgericht Walsrode • HRB 120986
    Ust-Id: DE191468481 • St.Nr.: 36/211/08000
    GPG Public Key zur Signaturprüfung:
    http://www.automatix.de/juergen_sauer_publickey.gpg

    —